[gst-devel] RTP plugins' status

[Zeeshan Ali]

Hello All,

    So lets summerise our work on GstRtp plugins, which we currently base on gphone's rtp implementation named librtp. The rtpsend was first written by zaheer, but there was no rtprecv. So i & wtay started working on it & soon realized that rtpsend also needs a lot of change. So we made progress & now we have the rtp plugins streaming. But the things didnt end up so simple. We wanted to support gsm, mpeg audio/video( layer 1,2 & 3 ),raw audio etc. All of these seem to work nice now but the mpeg audio layer 3( i havent got the chance to check layer 1 & 2 though ), which uses some forward refrences in its algorithem or something. 

I added a packet-sequencing algorithem of mine too in the rtprecv that works but i think there is some bug in it too.

I was busy today making the mtu( maximum transfer unit ) in the rtpsend to change according to the payload type & tell the rtprecv about its current value along with all other things it already tell's rtprecv. I watched some 4-5 min. mpeg 1 videos with mtu=1024 today & it all worked fine. wtay: plz decide/change the mtu sizes accordingly in the newcaps() of rtpsend.


ZEESHAN ALI


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